Added difficulty: for testing, it is much easier to run everything on the same system.
Unfortunately, even when using the system's public network IP, the network subsystem will take a shortcut and route through the loopback device.
So you have to apply the rules there, ie:

tc qdisc add dev lo root netem delay 100ms 50ms 25%

And remember to remove them when you're done as this will interfere with lots of things..

Just as a note (as much so I will remember &/or be able to find more easily as for anyone else's benefit), I've managed to get some other tc functions to work as well: loss, reorder, delete all, and list active (examples above for an eth0 device).

At the moment, we detect the network bottleneck because the network write call takes longer to return, handling WSAEWOULDBLOCK and socket.timeout would be more explicit.
Maybe we shouldn't be using blocking sockets? Or maybe reads can be blocking but it would be useful if writes were not so that we could detect when the network layer cannot handle any more data. (assuming that we can distinguish)
Or maybe we need to use different code altogether for win32 and posix?

The big hurdle for fixing this is that we have a number of queues and threads sitting in between the window damage events and the network sockets.
When things go wrong (network bottleneck, dropped packets, whatever), we need to delay the window pixel capture instead of queuing things up downstream (pixel encoding queue, packet queue, etc).
These buffers were introduced to ensure that we keep the pixel pipeline filled at all times to make the best use of the available bandwidth: highest fps / quality possible.
When tweaking those settings, we want to make sure we don't break the optimal use case.
So maybe we should define a baseline before making any changes, one for the optimal use case (gigabit or local connection without mmap) and one for the "slow" network connection (fixed settings we can reproduce reliably with tc). That's on top of the automated perf tests, which will give us another angle on this.

Things to figure out:

XPRA_BATCH_ALWAYS=1 - I don't think this would make much of a difference, we always end up batching anyway - but worth checking

XPRA_MAX_SOFT_EXPIRED=0 - how much does this help? It will prevent us from optimistically expiring damage regions before we have received ACK packets. Not sure yet how we would tune this automatically

TARGET_LATENCY_TOLERANCE=0 (added in r15616) - how much difference does this make? (probably very little on its own, likely to require XPRA_MAX_SOFT_EXPIRED=0)

av-sync: does this even matter?

r15617 exposes the "damage.target-latency" for each window, how far is this value from the network latency? This needs to account for the client processing the packet, decoding the picture data and presenting it, sending a new packet.. so an extra ~50ms is to be expected.

The heuristics quickly adapt to hitting the ceiling in this case: it takes under a second to go back to no delay at all (probably thanks to the batch delay adjusting dynamically with the backlog).
We should be able to spot the first packet that takes a long time to push out and raise the batch delay immediately.
When the send speed is restricted, its value ends up being pretty close to the actual line constraints, so we could implement a soft bandwidth limit, detected at runtime.

And in the meantime, packets (and that's any packet, not just picture updates...) can take a second to send.. so the user experience suffers.
The difficulty here is that during those first ~120ms, we may have already encoded and queued another 6 packets...
We may get a signal in the form of a socket timeout or of partial send, but this would be caught very deep in the network layer, far from where we would be able to do anything about it.
So I have a patch that tries to deal with this by scheduling very short timers, that we cancel if the send completes in time. This timeout value is still TBD, and may vary from OS to OS... testing will tell.
This at least prevents things from getting completely out of control, but it does create some stuttering as it overcompensates. Synchronizing via the UI thread may not be the right thing to do here (too slow / expensive).
Also, it turns any form of congestion into an increase in batch delay, which may not be the right thing to do if the send queue is busy (could be another window), as this is too drastic.
Combining this approach with a soft bandwidth limit could smooth things out better.

try harder to avoid using png: when limiting the bandwidth to 1Mbps (see #417), a single medium size png frame can take 5 seconds to send! (~600KB)

need to adjust speed and quality auto-tuning with congestion events, and include (soft) bandwidth limit in the heuristics

move the encode thread pause code: it applies to damage packets from all window sources since they all share the same thread (generalize the code so source doesn't need to know about the damage sequence)

clipboard packets also use the encode thread and generally don't need throttling (and throttling them can cause problems.. priority queues?) - same for cleanup functions, those don't need to wait

ideally: get rid of closures from queue_damage_packet

some constants should be derived: how long we pause sending during congestion, how long a send is expected to take

we should support bandwidth-limit=none to turn off all congestion code, even detected bandwidth - without using env vars

try harder to stop queuing more packets when we already have late_packets

bandwidth budget: evaluate the compressed frame size before proceeding? (on very limited bandwidth setup, each frame may use 25% or more of the bandwidth, allowing it through when we get just under 100% of the budget may still be too much)

auto-refresh kicks in because we delay things so much that we never reach the code that re-schedules it... same for get_refresh_subregion_encoding

initial speed and quality should be lower when the bandwidth-limit is low

The big remaining problem: a single png frame can kill the bandwidth...
And we can trigger it from quite a few places. Even a jpeg at high quality can use up the whole budget for several seconds.
Then we also end up queuing too many screen updates. Maybe we can inspect the size of the queue and give up on it?
We should also switch to h264 more quickly.

I can now run glxspheres at 4 to 8fps on a 1Mbps connection, without getting any stuttering!

If anything, we use less bandwidth than we should (~250Kbps). Being under the limit is good, as it allows the odd large frame to go through without causing too many problems, but this means that the bandwidth-limit detection code never fires, and maybe we could raise the quality or speed (framerate) a bit.

For very limited bandwidth situations, it would be nice to have a more progressive picture refresh, either flif (#992) or just raising the jpeg / webp quality up, only doing true lossless if we have time and bandwidth.

With all the latest improvements and fixes (r17491, r17493, r17515, r17516, etc) and running glxspheres for testing, the server ends up turning on b-frames (#800), video scaling and YUV420P subsampling. It then reaches 25fps on a 1Mbps connection!

I think this is as good as it is going to get for this release. We can improve it some more in #1700.

@maxmylyn: bear in mind that a lossless refresh will use up a few seconds' worth of bandwidth, this is unavoidable. We try to delay the refresh as much as possible, especially when we know the bandwidth is limited - this works better if we know in advance that there are limits (#417).
It's a balancing act between quality, bandwidth usage and framerate.

We need results from the automated tests to ensure that the performance has not regressed for the non-bandwidth-constrained case.

When debugging, we need: "-d refresh,compress,regionrefresh", and the usual "xpra info". And a reproducible test case that does not involve user interaction, as I've launched glxspheres literally thousands of times to get here.

Some links for testing bandwidth constraints:

​netlimiter (win32) is quite easy to use, not sure how well it simulates network congestion

To try to mitigate the problems with the auto-refresh killing the bandwidth, r17540 switches to "almost-lossless" (using high quality lossy) when bandwidth is scarce.
Note: in some cases, webp lossless will compress better than jpeg lossy... it really depends on the type of content. #1699 may help with those decisions.

Okay, I did some maintenance today on the automated test box - unfortunately it's on Fedora 25, and will need to be upgraded sometime in the near future. I might be physically in office on Friday (strong maybe), so I can do it then. I don't want to risk doing an upgrade over the VPN because who knows what can go wrong.

Now that I'm not dealing with a horrible Fedora 25->26 upgrade I've had time to finally catch up on my tickets. While testing #1638 by watching some video files and reading Wikipedia at the same time (partly to pass the time, partly to add some extra strain on the session) I noticed that the session aggressively lowers quality to the point that it became almost impossible to read after scrolling or switching tabs unless I let the session sit for a couple seconds. I noticed the aggressive quality drops, but I thought I was going crazy when I thought my overall performance had tanked sometime last week - but the session was still responsive, so it wasn't lagging or drawing spinners to indicate a slow server. I went perusing through the timeline to see if anything had changed in the last couple weeks and I found the link to this ticket in r17540.

So, I rolled my server back to r17539 and rebuilt it, and re-launched a session - and lo and behold everything was pretty again. While the quality can drop if you aggressively scroll (as expected) generally the server refreshes much quicker and to the layman's eye the session looks and behaves much nicer.

So I guess I have a half question and a half response to this - would something in r17540 cause the server to aggressively pursue lower quality even when bandwidth wouldn't be scarce? My server is on a full gigabit (and fully wired, albeit kinda lengthy, Ethernet) LAN so there's no reason why it should think that it has limited bandwidth. Or am I completely wrong? (totally likely)

Also, unrelated, I'll be up physically at the test box to yet again run a Fedora upgrade to 26.

Any other later revision than r17539? There are ~68 revisions since then.

would something in r17540 cause the server to aggressively pursue lower quality

No.
It only switches to non-lossless auto-refresh if both of those conditions are met:

current quality is lower than 70%

there were congestion events recently, or the bandwidth-limit (measured or specified) is lower than 1Mbps

This would manifest itself as an auto-refresh that is lossy but near lossless, this does not affect how quickly it kicks in.

It does nothing to the "quality" setting, and AFAICR - the only changes to the quality setting calculations only apply when there are bandwidth constraints (r17456).

You didn't include any debug information (ie: "xpra info" would be a start, also "-d refresh,regionrefresh" since this a refresh issue), so we can't say for sure if this is at play here.

The changesets that do change how we handle auto-refresh are: r17491, r17481, r17480, r17458, r17456.
In particular, r17481+r17480 change when we cancel and schedule the auto-refresh, to ensure that we only schedule one after we have sent a lossy packet (and not just at the time the screen update is queued up).

Well, this ticket has been set to "critical" and was assigned to you for the 2.2 release, that would be a more obvious way.

Sorry about that - I've been finally able to catch up to my tickets late last week. Now that the holidays are approaching I'll definitely be able to get more done, especially after next week when I'm back full time.

Anyways:

No.
It only switches to non-lossless auto-refresh if both of those conditions are met:
...

It does nothing to the "quality" setting, and AFAICR - the only changes to the quality setting calculations only apply when there are bandwidth constraints (r17456).

Okay I had a feeling that I was mistaken but wanted to make sure, thanks for confirming that.

So I did what I should have done yesterday and turned on the OpenGL paint boxes to investigate what's actually going on rather than blindly guessing. What I'm seeing is that as of r17607 it's painting an entire Firefox window with h264 (that is still the blue one, right?) under certain situations. In certain situations it makes - like when I switch from tab to tab rather quickly or click through a series of Wikipedia links quickly. But there are other cases where it doesn't make sense - the two most notably:

Hovering over the citations on Wikipedia pages (causes a small pop up to appear) causes rather large h264 paints - in some cases full window

Clicking on a different window and back to Firefox causes a full window refresh with h264

After some further investigation - it's much easier to repro what I'm seeing by opening an Xterm and running for i in {1..500}; do dmesg; done and then let the Xterm sit for a second or two. With the OpenGL paint boxes (and the refresh logs running) it becomes obvious to me that it's the autorefresh that's painting as lossy.

So before I walk through the revisions you listed, try a few other things, and then get some logs / Xpra info, do you want to create a new ticket or is that related enough to this one to continue here? (In case you're still awake, but I think that's highly unlikely)

I've tried all the revisions you listed and more testing is inconclusive. I can't help but wonder if maybe it's the new Firefox update that's triggering what I'm seeing much more.

The first step, as always and as noted in comment:27, is xpra info, and also -d refresh,regionrefresh.
At least then we'll know what the refresh delay is, quality and so on.

It sounds to me like the heavy scrolling is triggering a video region covering most of the window, that's expected - that's how we trigger scrolling detection.
What should be happening is that the auto-refresh should be kicking in quickly, and maybe it isn't.
If there are minor screen updates within the video region, this can delay things further. Maybe we can detect those and refresh before timing out the video region.

The Automated Test Box upgrade was succesfull - unfortunately it hasn't been properly running the tests for a couple weeks due to Fedora 25 going EOL and it losing access to the 1.X packages in the Beta repo - I had a script that would curl the repo list and parse it for the latest 1.X and remove Xpra and install the latest 1.X and later upgrade to the latest 2.X before the next round of testing. Anyways, now that it's up to Fedora 26 it should continue with the tests as you originally asked in comment:23.

Also - you mention bandwidth constrained tests - I have some test runs with TC enabled but they add jitter and packet loss - I'll have to add a test case or two (maybe three?) that adds some hard bandwidth limits as well to see how that impacts us.

Anyways,

I got an xpra info for you of when it gets blurry before the auto-refresh kicks in several seconds later. I also managed to get some -d refresh,regionrefresh logs of my Xterm test case where I got it to stick blurry for quite some time before a timed autorefresh finally kicked in. I'll attach them shortly.

So, somehow the heuristics end up thinking that you only have ~56Kbps of bandwidth available. And somehow I doubt you're using a 56K modem.
What's odd is that the xpra info you've attached does not match this:

Was the "xpra info" captured at the same time as the log sample?
Congestion events that trigger bandwidth-limit calculations can be seen with "-d stats". This is what this ticket is about, if we get that wrong... we'll be driving the heuristics using bad data.

r17622 makes it possible to turn off "bandwidth-limit" detection with XPRA_BANDWIDTH_DETECTION=0 xpra start .., so at least you can use it to figure out if this is what is causing the slow refresh.

I simplified my setup to just two Xterms - one I had xpra info :13 ready to go - and in the other I ran for i in {1..500}; do dmesg; done because that reliably triggers the blurry state. What I did was run the dmesg Xterm and while it was still sitting at blurry I ran the xpra info. Much easier than firing up Firefox and playing around with it until it starts getting blurry.

Apologies for not including that in the first place. I'll attach the new log and xpra info.

But also I found that when running with XPRA_BANDWIDTH_DETECTION=0 the sound sync starts to go way off to the point where sound starts getting at least 500ms behind video after having a session run for a while - it starts out okay but after a few minutes the sound starts to get further and further behind. When running with XPRA_BANDWIDTH_DETECTION=1 sound sync is almost perfect.

EDIT: After the fact I've found that starting a YouTube video in Firefox and pausing video for several minutes (5 is the minimum to make it most notable) before hitting play again is where it's most notable that the sound starts to fall behind. Same thing applies to local video files so you don't need to fire up a Firefox / Google Chrome

So here's what I think is happening in this particular example: lots of very small screen updates (one character at a time: 12x13 pixels, less than 1KB) are followed by large full screen updates (499x316 or 480x312 pixels, 8KB minimum, climbing to 200KB).
The large packets eventually trigger the network throttling and we detect that event (record_congestion_event in the logs).
When we then calculate the bandwidth available, we ignore the last second or so to ensure that we don't count the packets that caused us to go over the limit, and so we only count the very small packets and end up calculating a very low bandwidth limit value.r17638 should mitigate that.

What I still don't understand is why your network setup starts pushing back after less than 150KB sent.
I'm not seeing that with either MS Windows or Fedora connecting through a very cheap 100Mbps switch.

This example was also very different from the original browser test case from comment:26.
Not sure at all how that one will fare.
If problems persist, please make sure to attach -d stats,refresh,refreshregion AND xpra info captured at the time when refresh problems occur.

What I still don't understand is why your network setup starts pushing back after less than 150KB sent.
I'm not seeing that with either MS Windows or Fedora connecting through a very cheap 100Mbps switch.

The house I live in wasn't wired for ethernet so we had to hand wire it in a ....very creative matter. I have to go through two gigabit switches (and around the entire house, in fact) to get to the crappy AT&T Gateway in the other room, so there's that. That being said I can easily consistently push way higher speeds than 56k - when doing a file transfer over the network it can easily push 150+mbits.

But, now that this has been sorted - sound sync has started to regress - I'm seeing it consistently behind by 300-500ms after a session has been running for a few minutes. Should I re-open #1638 or do you want to sort it here?

Applied to v2.2 in r17645.
What is the new detected bandwidth-limit value?
Is the refresh performance as good as it was in 2.1?

sound sync has started to regress

Sound-sync is not directly related to this.
Do you have an easily reproducible test case? Can you use it to bisect?
Are you confident that this regression is caused by this ticket?

The new bandwidth handling code can slow down screen updates, and this should not be a problem for sound sync which is usually behind the video anyway. (I assume you mean the audio is falling behind the video here, right?)
The new code should prevent us from swamping the connection, which should allow the audio packets to flow more reliably and prevent the kinds of bursts that cause the queue levels to rise.
We now also trigger video more quickly, which should help sound sync not hinder it.

The only downside that I can think of would be if the video framerate slows down so much that we end up not using a video encoding, as only video is synced. You can verify that with the paint boxes.

I'm seeing it consistently behind by 300-500ms after a session has been running for a few minutes.

Okay, back from vacation and now setting in to full time - this is the first ticket. I've ran it with the cutter disabled and that appears to be the culprit - I don't see a difference with and without the bandwidth detection enabled. So, I'll be closing this one (as it is not the cause and re-opening #1638 in a moment.

But before I actually do close this one I have one last question - is there a way to override the auto-detection to set it to a value higher or lower than what it would detect? It would be useful for testing.

Please specify the bandwidth configuration. I assume this is on a LAN? 100Mbps? With latency under 5ms?

wrt "rapid drop in quality":
Looking at the log file, minor congestion events (those can happen) seem to trigger a chain reaction: the latency tolerance goes down and we calculate a higher packets backlog (because we try to prevent flooding the pipe as soon as we think something might be going wrong), which triggers yet more soft congestion events..
So r18454 should handle this better, only triggering congestion events from the "soft expired" timer if those are new events.

For reference my network is a gigabit network - I have three Gigabit Netgear switches between my machine and the main Gateway (have to chain between the office and my room to get to the Gateway because the house isn't wired for Ethernet). It's a bit annoying but the network can reliably push large amounts of bandwidth - to the point where if I do a file transfer over Rsync or FTP I max out hard drive speeds.

I no longer get any warnings, but the bandwidth heuristics reliably drop down to a very low amount of bandwidth, which kills performance for me. I can get this to reliably happen by starting a session with Chrome and an Xterm and pull up a YouTube video and just sit and watch it for a few minutes - the framerate notably drops after a time and eventually becomes unwatchable. I'll retest this with VLC from a local file to make it easier to repro. In the meantime (installing VLC as I write this), I'll attach logs from my last session - they start out with a bandwidth limit of 17 megabytes/s which feels reasonable, but after a minute or so it drops this limit to 3 megabytes/s. I had to trim it as I let it run for several minutes while I made coffee and the file got really large - I have the untrimmed version if you want it. I trimmed it right at the point where it drops the limit to ~3 megabytes/s.

In another session I saw it drop down to under 1 megabytes/s while using FileZilla to SFTP a few video files over in a background virtual desktop (stuck even after switching back):

Is this a virtual machine or something? The other packets all arrive much quicker and this one is small.
We could (gradually) ignore the first few packets, just like we already ignore the first video frame, but I'm not sure this is warranted unless we see more of those during testing.

This is usually pretty reliable in bandwidth constrained setups, but maybe not in this case.
The log sample lasts 5 minutes, did the quality drop at the start? If not, when? Did it never recover?

Can you reproduce with "-d bandwidth,stats"?r18714 will only take "slow send" into account when we know that the bandwidth is already limited (default: less than 20Mbps) - if that helps, we'll need to make sure that we still detect congestion properly and bandwidth constrained setups.
Ideally, we should relax this rule and make sure that the quality goes back up instead. I will need logs for that.

I get low bandwidth warnings fairly regularly, and the limits almost immediately drop my bandwidth down to the minimum. I'll attach requested logs in a second or two.

In this log snap, I connect, and almost immediately the level drops and I never see the bandwidth go above ~3megabits. However towards the end all of a sudden the level increases and everything gets much clearer. Which is weird, but I'm glad I had the logs running to catch it. Hopefully it helps.

EDIT: In the log sample I'm attaching, I started a server with Firefox and an Xterm as start childs - I connected, opened a couple tabs and loaded up an HTML5 video in one and switched between it and a couple text sites a few times while the logs were running.

Before I give this back, I'd like to request the XPRA_BANDWIDTH_DETECTION and XPRA_MIN_BANDWIDTH flags get turned into control commands and/or at the very least arguments that can be fed into an xpra start. I'm sure there are lots of people who would like to either turn the heuristics off or at the very least leave it on auto but set it to a higher minimum on the fly. I know that there are a few people here who would definitely be interested.

... the limits almost immediately drop my bandwidth down to the minimum ...

Bugs squashed:

r18999: this would have caused the bandwidth detection to go to the minimum value instead of starting disabled. 1Mbps! no wonder you were seeing all sorts of problems

r19000 prevented limits from being raised again once the congestion subsided

Please try again.

I'd like to request ...

Let's first iron out the bugs before deciding what should and should not be configurable, and where. Environment variables are suitable places for the more obscure configuration options, ideally bandwidth detection can be made reliable enough that those options can remain hidden.
FYI, they can already be fed to xpra start using:

The pushback I'm seeing is the heuristics withhold a lot of the initial bandwidth for the first couple of seconds. Upon initial connection, everything is blurry and slow to respond (as is typical during network hiccups), and bandwidth monitors show it consistently using <1-2 mbps. Until a few seconds later.

I'll attach a log, but I'm not of the opinion that this behavior is entirely undesirable. It might not be a bad idea to put some small limits on the initial connection if the network can't handle that much traffic that instantaneously. I think we might be better served by slightly upping the min-bandwidth default. But I'll leave that decision to you.

For reference: I've upped my systems to r19184. However, I'm still running Fedora 26, I should carve out a day very soon to upgrade.

So r19224 increases the ack tolerance for the first few seconds after the client connects, or after a window reset.
This should prevent those relatively small screen updates from skewing the congestion detection.

Related: r19223 also raises the minimum bandwidth we detect from just 1Mbps to 5Mbps, most users should have at least that nowadays, right?
This one is unlikely to help with the initial bandwidth though: it kicks in when we detect bandwidth congestion and when things just start we don't have enough data yet to really trigger it.

I've been playing around with this for the last couple days (mostly using Xpra to write some documentation for something else entirely), and the changes you mentioned in comment:58 have wrapped up the last issue. As far as I'm concerned, this ticket is pretty much done. I'm gonna pass to you in case there's anything left you want to do.